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Timothy Smith

2009-May-16 11:46 UTC

[asterisk-users] Fwd: Asterisk With Cisco Voice Router

Hi,

In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware ?as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company (which will be the same as phone DID), Voice
mail, Auto Answer etc yet we need thousands of dollars to add those to
our cisco call manager 4.1 set up.

I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!

Has anyone had the above set up working successfully? Attached are some confs.

Thanks a lot for your assistance.

Kind Regards,
Wilson
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cs-intranet*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
103                        172.17.3.249                5060     OK (3 ms)
102                        172.17.3.248                5060     OK (3 ms)
101                        172.17.10.150               5060     OK (1 ms)
100/100                    172.19.4.102     D   N      32544    Unmonitored
4 sip peers [Monitored: 3 online, 0 offline Unmonitored: 1 online, 0 offline]


; 102 and 103 are cisco routers, 101 is the call manager, 100 is a SIP phone
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Steve Howes

2009-May-16 12:25 UTC

head link

[asterisk-users] Fwd: Asterisk With Cisco Voice Router

On 16 May 2009, at 12:46, Timothy Smith wrote:> <blah>
>
> Has anyone had the above set up working successfully? Attached are  
> some confs.
>
> Thanks a lot for your assistance.
Check about the sip.conf 'insecure' option. I have had to use it in  
the past for similar stuff. I think it was 'insecure=very' but that  
might be deprecated by now..

David Backeberg

2009-May-16 13:02 UTC

head link

[asterisk-users] Fwd: Asterisk With Cisco Voice Router

On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith at gmail.com>
wrote:> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
> and also a dialpeer to forward on the router to forward calls to my
> asterisk. It works properly but the problem is there is NO AUDIO! I
> have tried to change codec but no sucess!
> Has anyone had the above set up working successfully?
Yes.

You have been caught by a not-very-well-documented issue with setting
up voice routing on the 3845, and probably other similar Cisco gear.
And I'm not sure how you've done your test.
This is the closest I've ever seen to a document that explains your problem:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml

Did you have a SIP phone on one side of asterisk and a POTS phone on
the outside of the 3845?

If you did, and you could talk on both at the same time, I think you
would discover in fact that you do have some audio, in fact, one-way
audio to be precise. But I don't remember for sure, because it's been
a while since I've done this to myself.

At any rate, your problem is you have dial-peers to get voice packets
out from the 3845 to Cisco, but no dial-peers to get the packets from
SIP back to a physical circuit on the 3845. Think about this. What
should happen to a call inbound from asterisk, to the 3845? Should it
go out an E1 to the outside phones world? If so, you need to build a
dial-peer that does that. Until you do, you won't be getting two-way
audio.

you need another rule something like:
dial-peer voice 790792888 pots
map this back to the proper E1 circuit

A secondary problem could also be with the way you're managing your
DSPs. I don't know how many physical DSPs you have in your router, but
usually it's a GOOD thing to enable DSP farming.

Neeraj Chand

2009-May-17 16:38 UTC

head link

[asterisk-users] Fwd: Asterisk With Cisco Voice Router

Hi, 

We have AS5400's set up with asterisk boxes. Initially we had similar
issues, but as described, you need to have dial peers to handle both
incoming and outgoing peers.

Please post your dial peer configs as well as the serial interface
configs. I also found that until I add [isdn incoming-voice modem ] I
could not get incoming calls on that serial interface to route to my *
box.



-----Original Message-----
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asterisk-users-request at lists.digium.com
Sent: Saturday, 16 May 2009 10:00 AM
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 58, Issue 40

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Today's Topics:

   1. Fwd: Asterisk With Cisco Voice Router (Timothy Smith)
   2. Re: Fwd: Asterisk With Cisco Voice Router (Steve Howes)
   3. Re: Fwd: Asterisk With Cisco Voice Router (Timothy Smith)
   4. Re: Fwd: Asterisk With Cisco Voice Router (David Backeberg)
   5. Re: meetme dies looking for conf-getconfno (sean darcy)
   6. howto set up persistent dynamic meetme (sean darcy)
   7. Agent-Login/out in 1.6 (David Anthony O Reilly)
   8. Agent-Login/out in 1.6 (David Anthony O Reilly)
   9. Re: Agent-Login/out in 1.6 (Stefan Reuter)
  10. Re: Fwd: Asterisk With Cisco Voice Router (Timothy Smith)
  11. Re: Agent-Login/out in 1.6 (Jim Dickenson)
  12. Re: howto set up persistent dynamic meetme (Tilghman Lesher)
  13. Re: Fwd: Asterisk With Cisco Voice Router (Philipp Kempgen)


----------------------------------------------------------------------

Message: 1
Date: Sat, 16 May 2009 14:46:27 +0300
From: Timothy Smith <timotsmith at gmail.com>
Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID:
	<416fc8170905160446r5815fd87m67e62506ad9ac031 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,

In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware ?as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company (which will be the same as phone DID), Voice
mail, Auto Answer etc yet we need thousands of dollars to add those to
our cisco call manager 4.1 set up.

I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!

Has anyone had the above set up working successfully? Attached are some
confs.

Thanks a lot for your assistance.

Kind Regards,
Wilson
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cs-intranet*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
103                        172.17.3.249                5060     OK (3
ms)
102                        172.17.3.248                5060     OK (3
ms)
101                        172.17.10.150               5060     OK (1
ms)
100/100                    172.19.4.102     D   N      32544
Unmonitored
4 sip peers [Monitored: 3 online, 0 offline Unmonitored: 1 online, 0
offline]


; 102 and 103 are cisco routers, 101 is the call manager, 100 is a SIP
phone
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------------------------------

Message: 2
Date: Sat, 16 May 2009 13:25:40 +0100
From: Steve Howes <steve at geekinter.net>
Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <659DC612-4035-4D7E-A73C-77B5A16D607D at geekinter.net>
Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes


On 16 May 2009, at 12:46, Timothy Smith wrote:> <blah>
>
> Has anyone had the above set up working successfully? Attached are  
> some confs.
>
> Thanks a lot for your assistance.
Check about the sip.conf 'insecure' option. I have had to use it in  
the past for similar stuff. I think it was 'insecure=very' but that  
might be deprecated by now..



------------------------------

Message: 3
Date: Sat, 16 May 2009 15:46:51 +0300
From: Timothy Smith <timotsmith at gmail.com>
Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID:
	<416fc8170905160546o6ca0e92fma6361525b244855 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Thanks Steve for this tip.

I have insecure=very is not yet deprecated. I have added it but still no
good.

I personally think the problem could be with the codecs. Any ideas?

I have attached some debug info.

Regards,
Tim

On Sat, May 16, 2009 at 3:25 PM, Steve Howes <steve at geekinter.net>
wrote:>
> On 16 May 2009, at 12:46, Timothy Smith wrote:
>> <blah>
>>
>> Has anyone had the above set up working successfully? Attached are
>> some confs.
>>
>> Thanks a lot for your assistance.
>
> Check about the sip.conf 'insecure' option. I have had to use it in
> the past for similar stuff. I think it was 'insecure=very' but that
> might be deprecated by now..
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> ? http://lists.digium.com/mailman/listinfo/asterisk-users
>-------------- next part --------------
VG2#
VG2#
VG2#
VG2#
VG2#
May 16 12:41:40.237: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8  callref 0x0C73
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98392
                Exclusive, Channel 18
        Calling Party Number i = 0x2183, '730230199'
                Plan:ISDN, Type:National
        Called Party Number i = 0xA1, '790792888'
                Plan:ISDN, Type:National
May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x33B3 calltype 2 CALL_INCOMING
May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: call_incoming: call_id
0x33B3, Guid = BCBB464BB098
VG2#
May 16 12:41:40.241: fb_get_reject_cause_code: ERROR cause_code NULL

May 16 12:41:40.245: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name
in GTD
May 16 12:41:40.245: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8
callref = 0x8C73
        Channel ID i = 0xA98392
                Exclusive, Channel 18
May 16 12:41:40.353: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name
in GTD
May 16 12:41:40.353: ISDN Se0/0/1:15 Q931: TX -> ALERTING pd = 8
callref = 0x8C73
VG2#
May 16 12:41:46.697: ISDN Se0/0/0:15 SERROR: isdn_get_name_from_gtd:
false ret
May 16 12:41:46.697: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name
in GTD
May 16 12:41:46.697: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8  callref
= 0x8C73
May 16 12:41:46.713: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8
callref = 0x0C73
May 16 12:41:46.713: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x33B3 calltype 2 CALL_PROGRESS
OULKLAVG2#
May 16 12:41:46.713: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now
connected to 730230199 N/A
VG2#
May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8
callref = 0x0C73
        Cause i = 0x8490 - Normal call clearing
May 16 12:41:52.373: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x33B3 calltype 2 CALL_DISC
May 16 12:41:52.373: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now
connected to 730230199 N/A
May 16 12:41:52.373: %ISDN-6-DISCONNECT: Interface Serial0/0/1:17
disconnected from 730230199 , call lasted 5 seconds
VG2#
May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8  callref
= 0x8C73
May 16 12:41:52.385: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x0C73
May 16 12:41:52.385: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x33B3 calltype 2 CALL_CLEARED
-------------- next part --------------
From: "730230199" <sip:730230199 at 172.19.3.150>;tag=as5f114784
To: <sip:100 at
172.19.4.102:32544;rinstance=e6a140ee2d1dee0f>;tag=ae700477
Contact: <sip:730230199 at 172.19.3.150>
Call-ID: 7beff1bd661329c643aa69ec43628adc at 172.19.3.150
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.9
Content-Length: 0


---
    -- SIP/100-00820520 answered SIP/172.17.3.248-007fc920
Audio is at 172.19.3.150 port 13312
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.17.3.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.17.3.248:5060;branch=z9hG4bK501204;received=172.17.3.248
From: <sip:730230199 at 172.17.3.248>;tag=D8FE7BF8-4CA
To: <sip:730232888 at 172.19.3.150>;tag=as0fb38dd9
Call-ID: 4A137712-414D11DE-9606C927-51AF51F3 at 172.17.3.248
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:730232888 at 172.19.3.150>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 544232458 544232458 IN IP4 172.19.3.150
s=Asterisk PBX 1.6.0.9
c=IN IP4 172.19.3.150
t=0 0
m=audio 13312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Packet2Packet bridging SIP/172.17.3.248-007fc920 and
SIP/100-00820520
cs-intranet*CLI>
<--- SIP read from UDP://172.17.3.248:62582 --->
ACK sip:730232888 at 172.19.3.150:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK511A76
From: <sip:730230199 at 172.17.3.248>;tag=D8FE7BF8-4CA
To: <sip:730232888 at 172.19.3.150>;tag=as0fb38dd9
Date: Sat, 16 May 2009 12:38:27 GMT
Call-ID: 4A137712-414D11DE-9606C927-51AF51F3 at 172.17.3.248
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
cs-intranet*CLI>

------------------------------

Message: 4
Date: Sat, 16 May 2009 09:02:46 -0400
From: David Backeberg <dbackeberg at gmail.com>
Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID:
	<3de056a30905160602x6002a7a0jc21e67a346338d2d at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith at gmail.com>
wrote:> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
> and also a dialpeer to forward on the router to forward calls to my
> asterisk. It works properly but the problem is there is NO AUDIO! I
> have tried to change codec but no sucess!
> Has anyone had the above set up working successfully?
Yes.

You have been caught by a not-very-well-documented issue with setting
up voice routing on the 3845, and probably other similar Cisco gear.
And I'm not sure how you've done your test.
This is the closest I've ever seen to a document that explains your
problem:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0
080147524.shtml

Did you have a SIP phone on one side of asterisk and a POTS phone on
the outside of the 3845?

If you did, and you could talk on both at the same time, I think you
would discover in fact that you do have some audio, in fact, one-way
audio to be precise. But I don't remember for sure, because it's been
a while since I've done this to myself.

At any rate, your problem is you have dial-peers to get voice packets
out from the 3845 to Cisco, but no dial-peers to get the packets from
SIP back to a physical circuit on the 3845. Think about this. What
should happen to a call inbound from asterisk, to the 3845? Should it
go out an E1 to the outside phones world? If so, you need to build a
dial-peer that does that. Until you do, you won't be getting two-way
audio.

you need another rule something like:
dial-peer voice 790792888 pots
map this back to the proper E1 circuit

A secondary problem could also be with the way you're managing your
DSPs. I don't know how many physical DSPs you have in your router, but
usually it's a GOOD thing to enable DSP farming.



------------------------------

Message: 5
Date: Sat, 16 May 2009 09:04:36 -0400
From: sean darcy <seandarcy2 at gmail.com>
Subject: Re: [asterisk-users] meetme dies looking for conf-getconfno
To: asterisk-users at lists.digium.com
Message-ID: <gumdl4$o5b$1 at ger.gmane.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

sean darcy wrote:> sean darcy wrote:
>> Mark Michelson wrote:
>>> sean darcy wrote:
>>>> Danny Nicholas wrote:
>>>>> You "lost" conf-getconfno.gsm . Asterisk is
trying to play that
file to let>>>>> you pick a conference number to use.  It goes in
/var/lib/asterisk/sounds.>>>>> Grep for it.
>>>>>
>>>>> -----Original Message-----
>>>>> From: asterisk-users-bounces at lists.digium.com
>>>>> [mailto:asterisk-users-bounces at lists.digium.com] On
Behalf Of sean
darcy>>>>> Sent: Friday, May 15, 2009 12:39 PM
>>>>> To: asterisk-users at lists.digium.com
>>>>> Subject: [asterisk-users] meetme dies looking for
conf-getconfno
>>>>>
>>>>> With 1.6.1, I'm trying to set up a test of meetme for
creating
dynamic >>>>> conferences.
>>>>>
>>>>> cat meetme.conf
>>>>> [rooms]
>>>>> conf => 600
>>>>>
>>>>> extensions.conf:
>>>>>
>>>>> [meetme]
>>>>> exten => 2663,1,MeetMe(,D)
>>>>> exten => 2663,n,Hangup()
>>>>>
>>>>> exten => 2666,1,MeetMe()
>>>>> exten => 2666,n,Hangup()
>>>>>
>>>>>
>>>>> What I'm expecting is to dial 2663, get a conference
room number (
600, >>>>> I suppose since it's the only room ), and set a PIN.
>>>>>
>>>>> Then I'd dial 2666 enter the conference room number and
the PIN,
and be >>>>> put in conference.
>>>>>
>>>>> But here's what happens when I dial 2663:
>>>>>
>>>>>     -- Starting simple switch on 'DAHDI/1-1'
>>>>>      -- Executing [2663 at internal:1]
MeetMe("DAHDI/1-1", ",D") in
new stack>>>>> [2009-05-15 13:21:19] WARNING[2061]: file.c:641
ast_openstream_full: >>>>> File conf-getconfno does not exist in any format
>>>>> [2009-05-15 13:21:19] WARNING[2061]: file.c:924
ast_streamfile:
Unable >>>>> to open conf-getconfno (format 0x4 (ulaw)): No such file or
directory>>>>>    == Spawn extension (internal, 2663, 1) exited non-zero
on
'DAHDI/1-1'>>>>>      -- Hungup 'DAHDI/1-1'
>>>>>
>>>>>
>>>>> conf-getconfno does exist:
>>>>>
>>>>> ls -l /var/lib/asterisk/sounds/en/conf-getconf*
>>>>> -rw-r--r--. 1 root root 25211 2009-03-26 14:42 
>>>>> /var/lib/asterisk/sounds/en/conf-getconfno.ulaw
>>>>> -rw-r--r--. 1 root root 50466 2009-03-26 14:42 
>>>>> /var/lib/asterisk/sounds/en/conf-getconfno.wav
>>>>>
>>>>> Any help appreciated.
>>>>>
>>>>> sean
>>>>>
>>>> Will do, but why gsm? Nobody's using gsm, and it looks like
it's
seeking >>>> ulaw ( which is installed).
>>>>
>>>> sean
>>>>
>>> You may want to try setting languageprefix=yes in asterisk.conf in
the [options] >>> section. If that does not work, then you may wish to try to move
the
file up one >>> directory level and see if it plays, then.
>>>
>>> Mark Michelson
>>>
>> I did both:
>>
>> grep language asterisk.conf
>> languageprefix = yes ; Use the new sound prefix path syntax
>>
>> And I copied the files to /var/lib/asterisk/sounds - and then only
gsm >> and then only ulaw.
>>
>> Restarted. Rebooted.
>>
>> No luck.
>>
>> sean
>>
> 
> Filed as
> 
> https://issues.asterisk.org/view.php?id=15125
> 
> sean
> 
Well it works for me now if I enable [directories] in asterisk.conf.

sean




------------------------------

Message: 6
Date: Sat, 16 May 2009 09:21:43 -0400
From: sean darcy <seandarcy2 at gmail.com>
Subject: [asterisk-users] howto set up persistent dynamic meetme
To: asterisk-users at lists.digium.com
Message-ID: <gumela$qs6$1 at ger.gmane.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.

extensions.conf:

[meetme]
exten => 2663,1,MeetMe(,De)
exten => 2663,n,Hangup()

exten => 2666,1,MeetMe()
exten => 2666,n,Hangup()


What I'm expecting is to dial 2663, get a conference room number ( 600,
I suppose since it's the only room ), and set a PIN. Hangup.

Then users would dial 2666 enter the conference room number and the PIN,

and be placed in conference.

The problem is that as soon as 2663 hangs up, the conference disappears.

What I'd like is to have the conference stay up for some period of time 
after 2663 hangs up. That way I could schedule a conference for a 
specific time, set it up beforehand, and email everyone the conference 
number and pin.

I don't want the conference to stay up forever, since I'd like new
pin's

each time.

This should be a common use case. How do you do it?

sean




------------------------------

Message: 7
Date: Sat, 16 May 2009 14:49:36 +0100
From: David Anthony O Reilly <oreillda at tcd.ie>
Subject: [asterisk-users] Agent-Login/out in 1.6
To: cursor at telecomabmex.com
Cc: asterisk-users at lists.digium.com
Message-ID:
	<f81bfd510905160649v281ad769pc2a65785f2553cc0 at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Hi Carlos

"
       Agentcallbacklogin was deprecated in Asterisk 1.4 and eliminated
from
1.6 so you now need to use Dynamic Agents.  Although they claim that is
is simple enough to replace that functionality with dial plan code I
have yet to see a one line example that replaces everything the
agentcallbacklogin command did.|

I totally agree, I have never seen any example that makes it work. If
somebody shows me how to do it without using Voicemail I will let you
know.

Thanks
David

-- 
_________________________________________

Mr. David Anthony O'Reilly, B.Sc Comp (Hons)

M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) -
2009

Computer Science Graduate of The University of Dublin, Trinity College -
B.Sc (Comp) 2008

Email: oreillda at tcd.ie    /    dor3 at student.cs.ucc.ie
Tel: +353 (0) 86 030 60 32
_________________________________________
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Message: 8
Date: Sat, 16 May 2009 14:54:00 +0100
From: David Anthony O Reilly <oreillda at tcd.ie>
Subject: [asterisk-users] Agent-Login/out in 1.6
To: dickenson at cfmc.com
Cc: asterisk-users at lists.digium.com
Message-ID:
	<f81bfd510905160654s421ccc7am1143414b0d6009c7 at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Hi Jim

Thanks for your code!! I see you use the Voicemail system to
authenticate,
have you ever managed to avoid that as I don't use voicemail at all and
I am
thinking if I use that solution I will need to set up a voicemail for
all
the queue members just to get them to log in.

hehe What were the developers thinking by removing the old system! It
worked
perfect!! and by the looks of it nobody has ever recovered from the
command
removal unless they hack around with the voicemail system.

Hopefully somebody out there has managed to create an agent login/logout
without bringing voicemail into it???? If I find a way I will let you
and
post a wiki on it as I am sure loads of people have this problem.

Thanks
Dave


; #### Agent login logout ####
exten => *20,1,Answer()
exten => *20,n,wait(.0.5)
exten => *20,n,Read(AgentNumber,agent-user)
exten => *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten => *20,n,GotoIf($["${UserID}"=""]?NOUSER)
exten => *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten => *20,n,GotoIf($["${AgentStatus}"="1"]?VERIFY)
exten => *20,n,GotoIf($["${AgentStatus}"="2"]?VERIFY)
exten => *20,n(NOUSER),Playback(cfmc/bad-agent)
exten => *20,n,Playback(vm-goodbye)
exten => *20,n,Hangup()
exten => *20,n(VERIFY),VMAuthenticate(${AgentNumber}@ourvm)
exten => *20,n,GotoIf($["${AgentStatus}"="2"]?AGENTOFF)
exten => *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten => *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten =>
*20,n,AddQueueMember(support,Local/Queue${AgentNumber}@ansqueue,,,,${CUT
(CHA
NNEL,-,1)})
;   AQMSTATUS can be  ADDED | MEMBERALREADY | NOSUCHQUEUE
exten => *20,n,Playback(agent-loginok)
exten => *20,n,Verbose(2,Agent ${AgentNumber} added
${DB(users/${UserID}/AgentDevice)})
exten => *20,n,HangUp()
exten => *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten => *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten =>
*20,n,RemoveQueueMember(support,Local/Queue${AgentNumber}@ansqueue)
exten => *20,n,Playback(agent-loggedoff)
exten => *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten => *20,n,Hangup()"


-- 
_________________________________________

Mr. David Anthony O'Reilly, B.Sc Comp (Hons)

M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) -
2009

Computer Science Graduate of The University of Dublin, Trinity College -
B.Sc (Comp) 2008

Email: oreillda at tcd.ie    /    dor3 at student.cs.ucc.ie
Tel: +353 (0) 86 030 60 32
_________________________________________
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Message: 9
Date: Sat, 16 May 2009 16:06:04 +0200
From: Stefan Reuter <stefan.reuter at reucon.com>
Subject: Re: [asterisk-users] Agent-Login/out in 1.6
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <4A0EC84C.3050405 at reucon.com>
Content-Type: text/plain; charset="utf-8"

David Anthony O Reilly wrote:> hehe What were the developers thinking by removing the old system! It
> worked perfect!! and by the looks of it nobody has ever recovered from
> the command removal unless they hack around with the voicemail system.
I think the best solution is to either use an AGI script if agents
should be able to login/logout through the phone, this allows you to
store the agents in a database or any other external system.
On the other hand many call centers use special applications for their
agents and allow them to manage their availability through a GUI. Such
applications can make use of the Manager API. They probably more
convenient as they also allow to display additional information about
the calls they handle and the queues they are subscribed to.

=Stefan

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Message: 10
Date: Sat, 16 May 2009 17:22:22 +0300
From: Timothy Smith <timotsmith at gmail.com>
Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID:
	<416fc8170905160722n1e5d997dtcc244d827258fc3 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

David,

Thanks a lot for your input.  I will enable DSP farming. Like some
other techies, I just wanted to see it work before i consider others
things.

I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.

It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.

Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.

; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten => _X.,1,Read(NUM,beep,4,2,3)
exten => _X.,n,Dial(SIP/${NUM})

; This is all i get when i call and the call hangs up!

cs-intranet*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [730732888 at default:1]
Read("SIP/172.17.3.248-30069280", "NUM,beep,4,2,3") in new
stack
    -- Accepting a maximum of 4 digits.
  == Using SIP RTP CoS mark 5
    -- Executing [730732888 at default:1]
Read("SIP/172.17.3.248-30069280", "NUM,beep,4,2,3") in new
stack
    -- Accepting a maximum of 4 digits.
cs-intranet*CLI>

Thanks alot for your assistance.

On Sat, May 16, 2009 at 4:02 PM, David Backeberg <dbackeberg at gmail.com>
wrote:> On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith at
gmail.com>
wrote:>> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
>> and also a dialpeer to forward on the router to forward calls to my
>> asterisk. It works properly but the problem is there is NO AUDIO! I
>> have tried to change codec but no sucess!
>> Has anyone had the above set up working successfully?
>
> Yes.
>
> You have been caught by a not-very-well-documented issue with setting
> up voice routing on the 3845, and probably other similar Cisco gear.
> And I'm not sure how you've done your test.
> This is the closest I've ever seen to a document that explains your
problem:>http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0
080147524.shtml>
> Did you have a SIP phone on one side of asterisk and a POTS phone on
> the outside of the 3845?
>
> If you did, and you could talk on both at the same time, I think you
> would discover in fact that you do have some audio, in fact, one-way
> audio to be precise. But I don't remember for sure, because it's
been
> a while since I've done this to myself.
>
> At any rate, your problem is you have dial-peers to get voice packets
> out from the 3845 to Cisco, but no dial-peers to get the packets from
> SIP back to a physical circuit on the 3845. Think about this. What
> should happen to a call inbound from asterisk, to the 3845? Should it
> go out an E1 to the outside phones world? If so, you need to build a
> dial-peer that does that. Until you do, you won't be getting two-way
> audio.
>
> you need another rule something like:
> dial-peer voice 790792888 pots
> map this back to the proper E1 circuit
>
> A secondary problem could also be with the way you're managing your
> DSPs. I don't know how many physical DSPs you have in your router, but
> usually it's a GOOD thing to enable DSP farming.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> ? http://lists.digium.com/mailman/listinfo/asterisk-users
>-------------- next part --------------
May 16 14:18:28.640: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x0CAB
        Cause i = 0x80D1 - Invalid call reference value
May 16 14:18:28.640: ISDN Se0/0/1:15 SERROR: L3_GetUser_NLCB: EVENT 0X5A
No NLCB 2
May 16 14:18:28.640: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event
0x5A cr 0x8CAB callid 0x0
May 16 14:18:28.660: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8  callref 0x0CAC
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
        Calling Party Number i = 0x2183, '730730199'
                Plan:ISDN, Type:National
        Called Party Number i = 0xA1, '730732888'
                Plan:ISDN, Type:National
May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x3407 calltype 2 CALL_INCOMING
May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: call_incoming: call_id
0x3407, Guid = 42D2FC70B0D1
May 16 14:18:28.664: fb_get_reject_cause_code: ERROR cause_code NULL

May 16 14:18:28.668: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name
in GTD
May 16 14:18:28.668: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8
callref = 0x8CAC
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
May 16 14:18:28.676: ISDN Se0/0/0:15 SERROR: isdn_get_name_from_gtd:
false ret
May 16 14:18:28.676: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name
in GTD
May 16 14:18:28.676: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8  callref
= 0x8CAC
May 16 14:18:28.688: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8
callref = 0x0CAC
May 16 14:18:28.688: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x3407 calltype 2 CALL_PROGRESS
May 16 14:18:28.688: %ISDN-6-CONNECT: Interface Serial0/0/1:12 is now
connected to 730730199 N/A
May 16 14:18:28.788: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8
callref = 0x0CAC
        Cause i = 0x82AF - Resource unavailable, unspecified
May 16 14:18:28.788: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x3407 calltype 2 CALL_DISC
May 16 14:18:28.788: %ISDN-6-CONNECT: Interface Serial0/0/1:12 is now
connected to 730730199 N/A
May 16 14:18:28.788: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8  callref
= 0x8CAC
May 16 14:18:28.792: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8  callref
= 0x0000
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
        Restart Indicator i = 0x80
May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x3407 calltype 2 CALL_DISC
May 16 14:18:28.792: ISDN Se0/0/1:15 SERROR: CCPRI_Go: call id 0x3407
event 0x57 No ccb Source->HOST
May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x3407 calltype 2 CALL_CLEARED
May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid
1/0x0 calltype 6 CHAN_STATUS
OULKLAVG2#
May 16 14:18:28.792: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8
callref = 0x8000
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
        Restart Indicator i = 0x80
May 16 14:18:28.800: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x0CAC
        Cause i = 0x80D1 - Invalid call reference value
May 16 14:18:28.800: ISDN Se0/0/1:15 SERROR: L3_GetUser_NLCB: EVENT 0X5A
No NLCB 2
May 16 14:18:28.800: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event
0x5A cr 0x8CAC callid 0x0
OULKLAVG2#

------------------------------

Message: 11
Date: Sat, 16 May 2009 07:27:32 -0700
From: Jim Dickenson <dickenson at cfmc.com>
Subject: Re: [asterisk-users] Agent-Login/out in 1.6
To: David Anthony O Reilly <oreillda at tcd.ie>
Cc: Asterisk User MailList <asterisk-users at lists.digium.com>
Message-ID: <C6341B64.13C7D0%dickenson at cfmc.com>
Content-Type: text/plain; charset="iso-8859-1"

Where do you want to get the value that agent uses to validate?

You can do your own code to do the validation.

Get the value from where ever and then do a read and compare the value
read
with the value you retrieved from where ever. If there is  match you are
done if no match say error, maybe setup some counter and ask for value
again.

Or else use

  Authenticate(password[,options[,maxdigits]]): This application asks
the
caller
to enter a given password in order to continue dialplan execution. If
the
password
begins with the '/' character, it is interpreted as a file which
contains a
list of
valid passwords, listed 1 password per line in the file.
  When using a database key, the value associated with the key can be
anything.
Users have three attempts to authenticate before the channel is hung up.
  Options:
     a - Set the channels' account code to the password that is entered
     d - Interpret the given path as database key, not a literal file
     m - Interpret the given path as a file which contains a list of
account
         codes and password hashes delimited with ':', listed one per
line
in
         the file. When one of the passwords is matched, the channel
will
have
         its account code set to the corresponding account code in the
file.
     r - Remove the database key upon successful entry (valid with 'd'
only)
     maxdigits  - maximum acceptable number of digits. Stops reading
after
         maxdigits have been entered (without requiring the user to
         press the '#' key).
         Defaults to 0 - no limit - wait for the user press the '#' key.

-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/




From: David Anthony O Reilly <oreillda at tcd.ie>
Date: Sat, 16 May 2009 14:54:00 +0100
To: <dickenson at cfmc.com>
Cc: <asterisk-users at lists.digium.com>
Subject: Agent-Login/out in 1.6

Hi Jim

Thanks for your code!! I see you use the Voicemail system to
authenticate,
have you ever managed to avoid that as I don't use voicemail at all and
I am
thinking if I use that solution I will need to set up a voicemail for
all
the queue members just to get them to log in.

hehe What were the developers thinking by removing the old system! It
worked
perfect!! and by the looks of it nobody has ever recovered from the
command
removal unless they hack around with the voicemail system.

Hopefully somebody out there has managed to create an agent login/logout
without bringing voicemail into it???? If I find a way I will let you
and
post a wiki on it as I am sure loads of people have this problem.

Thanks
Dave


; #### Agent login logout ####
exten => *20,1,Answer()
exten => *20,n,wait(.0.5)
exten => *20,n,Read(AgentNumber,agent-
user)
exten => *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten => *20,n,GotoIf($["${UserID}"=""]?NOUSER)
exten => *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten => *20,n,GotoIf($["${AgentStatus}"="1"]?VERIFY)
exten => *20,n,GotoIf($["${AgentStatus}"="2"]?VERIFY)
exten => *20,n(NOUSER),Playback(cfmc/bad-agent)
exten => *20,n,Playback(vm-goodbye)
exten => *20,n,Hangup()
exten => *20,n(VERIFY),VMAuthenticate(${AgentNumber}@ourvm)
exten => *20,n,GotoIf($["${AgentStatus}"="2"]?AGENTOFF)
exten => *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten => *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten =>
*20,n,AddQueueMember(support,Local/Queue${AgentNumber}@ansqueue,,,,${CUT
(CHA
NNEL,-,1)})
; ? AQMSTATUS can be ?ADDED | MEMBERALREADY | NOSUCHQUEUE
exten => *20,n,Playback(agent-loginok)
exten => *20,n,Verbose(2,Agent ${AgentNumber} added
${DB(users/${UserID}/AgentDevice)})
exten => *20,n,HangUp()
exten => *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten => *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten =>
*20,n,RemoveQueueMember(support,Local/Queue${AgentNumber}@ansqueue)
exten => *20,n,Playback(agent-loggedoff)
exten => *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten => *20,n,Hangup()"


-- 
_________________________________________

Mr. David Anthony O'Reilly, B.Sc Comp (Hons)

M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) -
2009

Computer Science Graduate of The University of Dublin, Trinity College -
B.Sc (Comp) 2008

Email: oreillda at tcd.ie ? ?/ ? ?dor3 at student.cs.ucc.ie
Tel: +353 (0) 86 030 60 32
_________________________________________


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Message: 12
Date: Sat, 16 May 2009 10:17:54 -0500
From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
Subject: Re: [asterisk-users] howto set up persistent dynamic meetme
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <200905161017.54997.tilghman at mail.jeffandtilghman.com>
Content-Type: text/plain;  charset="iso-8859-1"

On Saturday 16 May 2009 08:21:43 sean darcy wrote:> With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
> conferences.
>
> extensions.conf:
>
> [meetme]
> exten => 2663,1,MeetMe(,De)
> exten => 2663,n,Hangup()
>
> exten => 2666,1,MeetMe()
> exten => 2666,n,Hangup()
>
>
> What I'm expecting is to dial 2663, get a conference room number (
600,> I suppose since it's the only room ), and set a PIN. Hangup.
>
> Then users would dial 2666 enter the conference room number and the
PIN,> and be placed in conference.
>
> The problem is that as soon as 2663 hangs up, the conference
disappears.> What I'd like is to have the conference stay up for some period of
time> after 2663 hangs up. That way I could schedule a conference for a
> specific time, set it up beforehand, and email everyone the conference
> number and pin.
>
> I don't want the conference to stay up forever, since I'd like new
pin's> each time.
>
> This should be a common use case. How do you do it?
In Asterisk 1.6, user DEA contributed realtime capabilities to MeetMe,
which
allows the capability of scheduling conferences, with new pins each
time.  I
believe this would meet the needs your question has posed.

-- 
Tilghman



------------------------------

Message: 13
Date: Sat, 16 May 2009 17:51:39 +0200
From: Philipp Kempgen <philipp.kempgen at amooma.de>
Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <4A0EE10B.1080003 at amooma.de>
Content-Type: text/plain; charset=ISO-8859-1

Steve Howes schrieb:> Check about the sip.conf 'insecure' option. I have had to use it in
> the past for similar stuff. I think it was 'insecure=very' but that
> might be deprecated by now..
insecure=very should now be written as insecure=port,invite


    Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 



------------------------------

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

End of asterisk-users Digest, Vol 58, Issue 40
**********************************************

Tarek Sawah

2009-May-26 22:41 UTC

head link

[asterisk-users] Fwd: Asterisk With Cisco Voice Router

through a test .. i was able to send calls from Asterisk 1.4 to a PSTN number
through a cisco router with a channel bank.. Audio worked well..  i setup a dial
plan in asterisk to Dial(${EXTEN}@ciscoip)  and authorise the cisco router's
ip on the asterisk server and treat the calls comming from it like any other SIP
calls inside the server..


--
AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308





> Date: Sat, 16 May 2009 14:46:27 +0300
> From: timotsmith at gmail.com
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
> 
> Hi,
> 
> In our office, we're slowly migrating from a cisco call manager set up
> to asterisk. Problem is management doesn't want to buy any other
> hardware  as they had already invested a lot in cisco. The main cause
> of this is asterisk's added features like unique FAX number for
> everyone in the company (which will be the same as phone DID), Voice
> mail, Auto Answer etc yet we need thousands of dollars to add those to
> our cisco call manager 4.1 set up.
> 
> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
> and also a dialpeer to forward on the router to forward calls to my
> asterisk. It works properly but the problem is there is NO AUDIO! I
> have tried to change codec but no sucess!
> 
> Has anyone had the above set up working successfully? Attached are some
confs.
> 
> Thanks a lot for your assistance.
> 
> Kind Regards,
> Wilson
_________________________________________________________________
Insert movie times and more without leaving Hotmail?.
http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd1_052009
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